[asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?

Thu Aug 27 16:57:35 CDT 2015

Are you using this method of setting headers on PJSIP?

https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Function_PJSIP_HEADER

On Thu, Aug 27, 2015 at 4:54 PM, Dan Cropp <dan at amtelco.com> wrote:

> Thanks Scott.
>
>
>
> I was able to get the basic concept to run.
>
> However, it seems PJSIP INVITE for the Dial also does not support added
> headers.
>
>
>
> The Local channel dial plan did have the channel variable values.  I added
> them as SIP headers, then Dial(PJSIP/Agent).
>
> The INVITE for the Dial on PJSIP continues to not include the SIP Headers
> I added.
>
>
>
> For chan_sip, I have no problem with this.  Even the original Queue code I
> had includes the added SIP headers with it’s INVITE to the Agent.
>
>
>
>
>
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Scott Griepentrog
> *Sent:* Thursday, August 27, 2015 4:28 PM
>
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Is it possible to perform PJSIP Add
> Header prior to calling Queue and have it part of the INVITE packet?
>
>
>
> Local channels:
> http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeeperDialplan_id324598.html
>
>
>
> This explains adding members to queues, although it doesn't specifically
> provide an example of using local channels in a queue:
> http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/ACD_id289508.html
>
>
>
> Basically, read that book, and if you get stuck ask for help.
>
>
>
>
>
> On Thu, Aug 27, 2015 at 3:08 PM, Dan Cropp <dan at amtelco.com> wrote:
>
> Thanks Scott.
>
>
>
> I’m taking over for someone else’s code, so I must admit I’m still
> learning the Agent and Queue concepts.  Local channels are something I have
> not used either.  Would local channels essentially be an internal bridge?
>
>
>
> How would I
>
> “Register Local/number at agent in the queue on behalf of the agent (replace
> number with the agent's extension number)”
>
>
>
>
>
>
>
> *From:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Scott Griepentrog
> *Sent:* Thursday, August 27, 2015 1:57 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* Re: [asterisk-users] Is it possible to perform PJSIP Add
> Header prior to calling Queue and have it part of the INVITE packet?
>
>
>
> To add a header to the call leg that goes to the agent, try using a local
> channel to activate dialplan on the outbound call:
>
>
>
> Register Local/number at agent in the queue on behalf of the agent (replace
> number with the agent's extension number)
>
>
>
> In dialplan [agent], wild card match the number, add the header, and then
> Dial(PJSIP/{$EXTEN}) to send the call to the agent.
>
>
>
>
>
> On Thu, Aug 27, 2015 at 1:40 PM, Dan Cropp <dan at amtelco.com> wrote:
>
> I have a call coming in.
>
> I need to add a SIP Header to the channel.
>
> Then, I need to send the call to the Queue so it is sent to the Agent.
>
>
>
> The SIP header I added, I need to have appear in the INVITE sent to the
> Agent.
>
>
>
> It works in chan_sip.  I send the call to a macro which does…
>
> n,SIPAddHeader(My-Header-Name:${MY-HEADER-VALUE})
>
> n,Queue(${ARG2})
>
>
>
>
>
> In PJSIP , this doesn’t seem to work.  Is there any way to add custom
> PJSIP headers to be sent as part of the INVITE to the Agent?
>
> When I look at the code, it seems as though the INVITE doesn’t look for
> any custom headers to be included with the INVITE packet.  Is this correct?
>
>
>
> Have a great day!
>
> Dan
>
>
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>
> Scott Griepentrog
> Digium, Inc · Software Developer
> 445 Jan Davis Drive NW · Huntsville, AL 35806 · US
> direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090
> Check us out at: http://digium.com · http://asterisk.org
>
>
> --
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>
>
> --
>
> [image: Digium logo]
>
> Scott Griepentrog
> Digium, Inc · Software Developer
> 445 Jan Davis Drive NW · Huntsville, AL 35806 · US
> direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090
> Check us out at: http://digium.com · http://asterisk.org
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
> asterisk-users mailing list
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-- 
[image: Digium logo]
Scott Griepentrog
Digium, Inc · Software Developer
445 Jan Davis Drive NW · Huntsville, AL 35806 · US
direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090
Check us out at: http://digium.com · http://asterisk.org
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